1 files changed, 113 insertions, 0 deletions
diff --git a/src/lib/resampler.cc b/src/lib/resampler.cc
new file mode 100644
index 000000000..7bc933fd0
--- /dev/null
+++ b/src/lib/resampler.cc
@@ -0,0 +1,113 @@
+/*
+    Copyright (C) 2013 Carl Hetherington <cth@carlh.net>
+
+    This program is free software; you can redistribute it and/or modify
+    it under the terms of the GNU General Public License as published by
+    the Free Software Foundation; either version 2 of the License, or
+    (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU General Public License for more details.
+
+    You should have received a copy of the GNU General Public License
+    along with this program; if not, write to the Free Software
+    Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+
+*/
+
+extern "C" {
+#include "libavutil/channel_layout.h"
+}	
+#include "resampler.h"
+#include "audio_buffers.h"
+#include "exceptions.h"
+
+#include "i18n.h"
+
+using std::cout;
+using std::pair;
+using std::make_pair;
+using boost::shared_ptr;
+
+Resampler::Resampler (int in, int out, int channels)
+	: _in_rate (in)
+	, _out_rate (out)
+	, _channels (channels)
+{
+	/* We will be using planar float data when we call the
+	   resampler.  As far as I can see, the audio channel
+	   layout is not necessary for our purposes; it seems
+	   only to be used get the number of channels and
+	   decide if rematrixing is needed.  It won't be, since
+	   input and output layouts are the same.
+	*/
+
+	_swr_context = swr_alloc_set_opts (
+		0,
+		av_get_default_channel_layout (_channels),
+		AV_SAMPLE_FMT_FLTP,
+		_out_rate,
+		av_get_default_channel_layout (_channels),
+		AV_SAMPLE_FMT_FLTP,
+		_in_rate,
+		0, 0
+		);
+	
+	swr_init (_swr_context);
+}
+
+Resampler::~Resampler ()
+{
+	swr_free (&_swr_context);
+}
+
+pair<shared_ptr<const AudioBuffers>, AudioContent::Frame>
+Resampler::run (shared_ptr<const AudioBuffers> in, AudioContent::Frame frame)
+{
+	AudioContent::Frame const resamp_time = swr_next_pts (_swr_context, frame * _out_rate) / _in_rate;
+		
+	/* Compute the resampled frames count and add 32 for luck */
+	int const max_resampled_frames = ceil ((double) in->frames() * _out_rate / _in_rate) + 32;
+	shared_ptr<AudioBuffers> resampled (new AudioBuffers (_channels, max_resampled_frames));
+
+	int const resampled_frames = swr_convert (
+		_swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) in->data(), in->frames()
+		);
+	
+	if (resampled_frames < 0) {
+		throw EncodeError (_("could not run sample-rate converter"));
+	}
+	
+	resampled->set_frames (resampled_frames);
+	return make_pair (resampled, resamp_time);
+}	
+
+shared_ptr<const AudioBuffers>
+Resampler::flush ()
+{
+	shared_ptr<AudioBuffers> out (new AudioBuffers (_channels, 0));
+	int out_offset = 0;
+	int64_t const pass_size = 256;
+	shared_ptr<AudioBuffers> pass (new AudioBuffers (_channels, 256));
+
+	while (1) {
+		int const frames = swr_convert (_swr_context, (uint8_t **) pass->data(), pass_size, 0, 0);
+		
+		if (frames < 0) {
+			throw EncodeError (_("could not run sample-rate converter"));
+		}
+		
+		if (frames == 0) {
+			break;
+		}
+
+		out->ensure_size (out_offset + frames);
+		out->copy_from (pass.get(), frames, 0, out_offset);
+		out_offset += frames;
+		out->set_frames (out_offset);
+	}
+
+	return out;
+}